175 lines
7.2 KiB
Markdown
175 lines
7.2 KiB
Markdown
# Snapcast client for ESP32
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### Synchronous Multiroom audio streaming client for [Snapcast](https://github.com/badaix/snapcast) ported to ESP32
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## Feature list
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- Opus, FLAC and PCM decoding currently supported
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- Wifi setup from menuconfig or through espressif Android App "SoftAP Prov"
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- Auto connect to snapcast server on network
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- Buffers up to 800ms on Wroom modules
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- Buffers more then enough on Wrover modules
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- Multiroom sync delay controlled from Snapcast server (user has to ensure not to set this too high on the server)
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## Description
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I have continued the work from @badaix, @bridadan and @jorgenkraghjakobsen towards a ESP32 Snapcast
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client. Currently it support basic features like multiroom sync, network
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controlled volume and mute. For now it only support Opus, FLAC and PCM 16bit
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audio streams with sample rates up to 48Khz maybe more, I didn't test.
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Please check out the task list and feel free to fill in.
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I dropped the usage of ADF completely but copied stripped down, needed components to this project.
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This was necessary because ADF was using flac in closed source precompiled library
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which made it impossible to get good results for multiroom syncing. IDF's I2S driver was also copied
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to project's components and adapted. Originally it wasn't possible to pre load DMA buffers with audio
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samples and therefore no precise sync could be achieved.
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### Codebase
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The codebase is split into components and build on ESP-IDF v4.3. I still
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have some refactoring on the todo list as the concept has started to settle and
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allow for new features can be added in a structured manner. In the code you
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will find parts that are only partly related features and still not on the task
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list.
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Components
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- MerusAudio : Low level communication interface MA12070P
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- flac : flac audio cider/decoder full submodule
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- opus : Opus audio coder/decoder full submodule
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- rtprx : Alternative RTP audio client UDP low latency also opus based
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- lightsnapcast : o Port of @bridadan scapcast packages decode library
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o player module, which is responsible for sync and low level I2S control
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- libmedian: Median Filter implementation. Many thanks to @accabog https://github.com/accabog/MedianFilter
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- libbuffer : Generic buffer abstraction
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- esp-dsp : Submodule to the ESP-ADF done by David Douard
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- dsp_processor : Audio Processor
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- audio-board : taken from ADF, stripped down to strictly necessary parts for usage with Lyrat v4.3
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- audio-hal : taken from ADF, stripped down to strictly necessary parts for usage with Lyrat v4.3
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- audio-sal : taken from ADF, stripped down to strictly necessary parts for usage with Lyrat v4.3
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- esp-peripherals : taken from ADF, stripped down to strictly necessary parts for usage with Lyrat v4.3
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- custom-driver : modified I2S driver from IDF v4.3 which supports preloading DMA buffers with valid data
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The snapclient functionanlity are implemented in a task included in main - but
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will be refactored to a component in near future.
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Todo: following does not apply to this fork
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Sync concept has been changed start 2021 on this implementation and differ a
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bit from the way original snap clints handle this.
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The snapclient frontend handles communiction with the server and after
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successfully hello hand shake it dispatches packages from the server.
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- CODEC_HEADER : Setup client audio codec (FLAC, OPUS, OGG or PCM) bitrate, n
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channels and bits pr sample
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- WIRE_CHUNK : Coded audio data
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- SERVER_SETTING : Channel volume, mute state, playback delay etc
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- TIME : Ping pong time keeping packages to keep track of time dif from server
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to client
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Each wire_chunk of audio data comes with a timestamp and client has agreed play
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that sample playback-delay after the timestamp. One way to handle that is to
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pass on audio data to a buffer with a length that compensate for for
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playback-delay, network jitter and DAC to speaker.
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In this implementation I have separated the sync task to a backend on the other
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end of a large ring buffer. Now the front end just need to pass on the audio
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data to the ring buffer with the server timestamp and chunk size. The backen
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read timestamps and waits until the audio chunk has the correct playback-delay
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to be written to the DAC amplifer speaker pipeline. When the backend pipeline
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is in sync, any offset get rolled in by micro tuning the APLL on the ESP. No
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sample manipulation needed.
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### Hardware
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- ESP pinout MA12070P
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------------------------------------------------------
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-> I2S_BCK Audio Clock 3.072 MHz
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-> I2S_WS Frame Word Select or L/R
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-> GND Ground
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-> I2S_DI Audio data 24bits LSB first
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-> MCLK Master clk connect to I2S_BCK
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-> I2C_SCL I2C clock
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-> I2C_SDA I2C Data
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-> GND Ground
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-> NENABLE Amplifier Enable active low
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-> NMUTE Amplifier Mute active low
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## Build
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Clone this repo:
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git clone https://github.com/jorgenkraghjakobsen/snapclint
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Update third party code (opus and esp-dsp):
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git submodule update --init
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Configure to match your setup
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- Wifi network name and password
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- Audio coded setup
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idf.py menuconfig
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Build, compile and flash:
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idf.py build flash monitor
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## Test
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Setup a snapcast server on your network
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On a linux box:
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Clone snapcast build and start the server
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./snapserver
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Pipe some audio to the snapcast server fifo
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mplayer http://ice1.somafm.com/secretagent-128-aac -ao pcm:file=/tmp/snapfifo -af format=s16LE -srate 48000
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Test the server config on other knowen platform
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./snapclient from the snapcast repo
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Android : snapclient from the app play store
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## Contribute
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You are very welcome to help and provide [Pull
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Requests](https://docs.github.com/en/github/collaborating-with-issues-and-pull-requests/about-pull-requests)
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to the project.
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We strongly suggest you activate [pre-commit](https://pre-commit.com) hooks in
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this git repository before starting to hack and make commits.
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Assuming you have `pre-commit` installed on your machine (using `pip install
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pre-commit` or, on a debian-like system, `sudo apt install pre-commit`), type:
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```
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:~/snapclient$ pre-commit install
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pre-commit installed at .git/hooks/pre-commit
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```
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Then on every `git commit`, a few sanity/formatting checks will be performed.
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## Task list
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- [ok] Fix to alinge with above
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* kconfig
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* add codec description
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- [ ] Integrate ESP wifi provision
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- [ok] Find and connect to Avahi broadcasted Snapcast server name
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- [ ] Add a client command interface layer like volume/mute control
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- [ ] Build a ESP-ADF branch
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## Minor task
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- [ ] Propergate mute/unute from server message to DSP backend mute control.
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- [ ] soft mute - play sample in buffer with decreasing volume
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- [ok] hard mute - pass on zero at the DSP hackend
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- [ ] Startup: do not start parsing on samples to codec before sample ring buffer hits requested buffer size.
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- [ok] Start from empty buffer
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